similar to: R: Dect to SIP PCI card

Hi all, I'm looking for a PCI card which i could install on asterisk box, with purpose to use 15-20 cordless dect phone in a very "dect cell". Is there anyone that could help me pls ? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks Jorg, it's worked, but what modules i need to use it with asterisk? I insert load => chan_capi.so in /etc/asterisk/modules.conf and chan_capi.so=yes under [globals] section. When asterisk start, I get this error: == Parsing '/etc/asterisk/modules.conf': Found [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf':

I'm going to try, Thanks very much Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Remco Barende Inviato: luned? 30 gennaio 2006 20.04 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Kirk IP600 Hi! Yes, it works (sort of) but I still have some issues.

I have to re-compile also app_rxfax.so and app_txfax.so or just spandsp ? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 12 gennaio 2006 17.20 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users]

Hi Francois, I kwnow, but I have "DSP:on" also if i not have an hardware echocan module :/ and I always have "Echo Cancellation: 0 taps, currently OFF". This is my zapata.conf [channels] language = it usecallerid = yes callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes cancallforward = yes callreturn = yes switchtype = euroisdn

I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Alexander Lopez Inviato: venerd? 16 settembre 2005 17.53 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: RE: [Asterisk-Users] direct sip call

I did it Steve, but on some calls i still have the EC on OFF. What can i check? Could it depend of my zapata.conf ? Thanks Giordano Grandis -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Steve Davies Inviato: marted? 28 marzo 2006 17.08 A: Asterisk Users Mailing List - Non-Commercial Discussion

Thanks...which version of IPP did u use ? I do not have Makefile file....there is only a .sh script Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Juan Salas Inviato: luned? 3 ottobre 2005 15.41 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE:

Ok, but is there a way to check if echo cancellation is active on a call in progress ? Thanks Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Steve Davies Inviato: marted? 28 marzo 2006 16.43 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Echo cancellation

Perfect, thanks very much hth. I just set it to unknown, but it doesn't work. Have I to use also prilocaldialplan ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 29 settembre 2005 16.22 A: 'Asterisk Users Mailing List -

Hi guys, just to let u known the pinout for the adapter : Adapter for connect the E1 telco lines on my digium card DB9 RJ45 3 1 8 2 2 4 6 5 Adapter used for connect the Hicom150 traditional PBX on my digium card (cross connection) DB9 RJ45 3 4 8 5 2 1

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of

Hi all, i'm tryinf to install chan_capi but i get this error root@obelix:/usr/src/chan_capi-0.3.5# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO

Patch it by hand. Follow this help http://www.asteriskguru.com/tutorials/spandsp.html Hi Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Tomislav Parcina Inviato: venerd? 13 gennaio 2006 12.28 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] RE: RE: Spandsp In article

Hi all, has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway? Could it works using the skinny channel ? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060130/07ec3c90/attachment.htm

Thanks Kristian, but i just answered to call, how can i use the Read application? Thanks Giordano Grandis -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Kristian Kielhofner Inviato: luned? 6 marzo 2006 18.15 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Capturing

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Which models of Diva could work with CAPI and asterisk? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di gw@adcomcorp.com Inviato: sabato 1 ottobre 2005 23.46 A: asterisk-users@lists.digium.com Oggetto: RE: [Asterisk-Users] Diva Nope. At least I tried and never could get it

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters